(1) Field of the Invention
The present invention is related to the field of recording and playback of audio information from hard drives and other related media, more specifically, the method and apparatus of the present invention is related to late buffer processing of audio information.
(2) Related Art
Audio processing including editing, playback and recording of audio is an important component of today""s multimedia and film related technology. Various factors may disrupt the normal processing of audio information.
One such factor is a delay in the audio processing which disrupts the normal flow of audio output in recording and playback of audio information. For example, if the audio information stored on disk is not read into memory in time, the wrong or obsolete audio information which was previously in memory may be reproduced. In prior art technology, such delay in audio processing has been largely ignored. Consequently, when the program processing audio information attempts to retrieve the audio information from memory, it is often the wrong audio information.
In other prior art technology, the audio processing demand on a system is decreased in an attempt to avoid delays in audio processing. More specifically, channel playback capacity is reduced in order to lower the amount of audio information being processed. For example, a playback recorder having eight audio channels may only allow utilization of three of its eight audio channels thereby reducing channel playback capacity. Since the system demand is only from three audio channels instead of the actually available eight audio channels, the risk that the system will slow down its audio processing is thought to be reduced.
There are many disadvantages related to the prior art technology. One such disadvantage is that the prior art technology does nothing to solve the problem of audio processing delay causing output of unrelated audio information. In such cases, audio information which has already been processed or incomprehensible audio is accessed by the system from memory reproducing unrelated and sometimes unpleasant audio output. Such systems also have no mechanisms for recovering after the system has caught up with its audio processing.
Yet another disadvantage of the prior art technology is the disruption of synchronization caused between audio channels. More specifically, synchronization is disrupted when one or more audio channels reproduce unrelated audio information where the lengths of such unrelated audio information is not equal to the corresponding audio information intended to be synchronously reproduced by other audio channels in the same system.
It is therefore desirable to have a method and apparatus for performing late audio buffer processing without causing disruptions to audio channel synchronization and preventing access to obsolete audio information.
The present invention discloses a method and apparatus for intelligently muting audio output when audio buffers are not read fast enough from disk into digital signal processing (DSP) memory. Given a playback audio recorder, each channel of audio is associated with a forward and a reverse late threshold objects which are enabled when the processing of an audio buffer for that channel is delayed. When an audio buffer for that channel is delayed, the late threshold signal sets in motion the muting of that specific audio channel for a predetermined amount of time. If and when the system catches up, normal playback resumes. Otherwise, the channel stays muted.
A real time module (RTM) of a known period and resolution is utilized to accurately predict when audio buffers should be read from the disk. The RTM also detects when any of the audio buffers are going to be late. When such delay is detected by the RTM, the mute processing is activated.
The method and apparatus of the present invention is advantageous over the prior art technology in that whenever a delay is detected by the present invention, a silence buffer containing mute information is accessed. Access of a silence buffer continues until such time that the system actually catches up and is no longer in a late state. The silence buffer accessed is guaranteed to be as large as the largest possible audio buffer to cover late audio information of any size.
Because the silence buffer is able to compensate for late audio information of any size and the system is continually monitoring its audio information access state, any audio channel which goes into a late state never goes out of synchronization from the remaining audio channels in the system. In other words, the system continues to process audio buffers of the length equal to those being processed by associated audio channels. Therefore, when the audio channel in a late state recovers and has caught up in audio processing, the audio buffer immediately following the silence buffer is accessed at the same time as the associated audio channels access their audio buffers.